Satoru Emura

Enhanced frequency-domain adaptive algorithm for stereo echo cancellation

Highly cross-correlated input signals create the problem of slow convergence of misalignment in stereo echo cancellation even after undergoing non-linear preprocessing. We propose a new frequency-domain adaptive algorithm that improves the convergence rate by increasing the contribution of non-linearity in the adjustment vector. Computer simulation showed that it is effective when the non-linearity gain is small.

Stereo echo cancellation algorithm using adaptive update on the basis of enhanced input-signal vector

Stereo echo cancellation requires a fast converging adaptive algorithm because the stereo input signals are highly cross correlated and the convergence rate of the misalignment is slow even after preprocessing for unique identification of stereo echo paths. To speed up the convergence, we propose enhancing the contribution of the decorrelated components in the preprocessed input-signal vector to adaptive updates. The adaptive filter coefficients are updated on the basis of either a single or multiple past enhanced input-signal vectors.For a single-vector update, we show how this enhancement improves the convergence rate by analyzing the behavior of the filter coefficient error in the mean. For a two-past-vector update, simulation showed that the proposed enhancement leads to a faster decrease in misalignment than the corresponding conventional second-order affine projection algorithm while computational complexities are almost the same.

A method of coherence-based step-size control for robust stereo echo cancellation

A fast adaptive algorithm is required for stereo echo cancellation because the strong interchannel cross-correlation of stereo received signals leads to an ill-conditioned normal equation to be solved by the adaptive filter. An adaptive algorithm for this task should also be robust against disturbances such as near-end speech and near-end noise. We propose a coherence-based method of step-size control that provides robustness in stereo echo cancellation. Computer simulation demonstrates that the method is robust against near-end speech and noise, and is capable of quickly tracking changes in echo paths.

On design of sequential sensor fusion system

Most algorithms currently proposed for sensor integration implicitly assume that the signals from all sensors are synchronized, but this does not hold in real situations. The paper proposes to handle time-variant sensor combination systems by a sequence, and formulates a method to verify that the designed sequence is stable, together with a method to predict theoretically the performance of the designed sequence based on the mutual information rate. This enables a sequence of sensor combinations to be designed rationally.

A method for posterior frequency-domain multi-channel residual echo canceling

We propose a posterior frequency-domain multi-channel residual echo canceling method. As the number of reproduction channels of an echo canceller increases, its adaptive filter shows a slower convergence of the filter coefficient error, that is, large echo path mismatch. In this situation, the change in the human talker in a remote site produces an abrupt residual echo more frequently. The proposed method immediately cancels this abrupt residual echo by estimating it from loudspeaker-reproduced signals, adaptive filter output, and echo replica with a much shorter frame length compared with adaptive filter length. By taking the echo replica into account, this frequency-domain method can improve residual-echo estimates. Another feature of the proposed method is controlling the residual echo estimates according to the confidence interval of these estimates.

Sound field estimation using two spherical microphone arrays

We propose a method of estimating a sound field with two spherical microphone arrays (SMAs). This method estimates plane-wave expansion coefficients of the sound field by using sparse representation modeling in the frequency domain. The dictionary matrix we propose for this modeling achieves the integration of the measurements of two SMAs in a straightforward manner. The effectiveness of the proposed method was evaluated in simulations with computer-generated and measured impulse responses.

ℓ 1 -constrained MVDR-based selection of nonidentical directivities in microphone array

We propose a method for finding a good combination of nonidentical directivities for a microphone array, i.e., a combination that can achieve better noise suppression. To avoid checking all combinations of directivities, the method incrementally applies ℓ1-constrained minimum variance distortionless response beamforming to the coherently-focused power spectral density matrices of microphone array signals. The ℓ1 constraint selects microphones with greater importance by inducing a sparse filter for beamforming.

Active haptic perception and control of its fixation

A computational approach is applied to the problem of controlling fixation of haptic perception during general motion. Estimation of hardness and curvatures of the environment from a single pressure image is shown to be ill-posed, and an active observation is demonstrated making the estimation well-posed. Also proposed is an algorithm for haptic servoing for the active finger-pad to fix its high-resolution region on the surface of contact.

Wave-domain canceling of residual echo with subspace tracking

For full-duplex immersive audio communication with wave field reconstruction, efficient echo canceling is required, and wave-domain adaptive filtering techniques have been developed. We show that in a linear array case, the spatial transformation to the wave domain can be interpreted as delay-and-sum beamforming. Based on this interpretation, we propose to take into account the reflection of dominant reproduced waves extracted by subspace tracking in order to improve the performance of residual echo canceling.

Adaptive filtering exploiting band-limited property of sound field in the wave domain

Massive multichannel sound reproduction is necessary for highly immersive full-duplex communication. Efficient echo cancellation between massive loudspeakers and microphones is indispensable, and wave-domain adaptive filtering reduces the computational complexity drastically. We propose to further reduce this complexity by exploiting the band-limited property that the sound field has in the space-time-frequency representation. Experimental evaluations support the applicability of the proposed method.