Active noise control (ANC) over a sizeable space ideally requires uniformly distributed sensors and secondary sources, which limits the feasibility of practically realizing such systems. In this paper, we propose a direction of arrival (DOA) weighting algorithm for the adaptive filter update, which prioritizes residual error control with respect to the array geometry. Array geometries utilizing multiple horizontal rings, which are considered as more practical than spherical array geometries, are introduced into both sensors and secondary sources. Numerical simulations indicate that the proposed method using multiple-horizontal-ring arrays gives higher noise attenuation performance than the conventional method. The DOA weighting can be intuitively defined on the basis of the secondary source array geometry without any prior information of the primary noise field.

Active acoustic noise attenuation over a sizable space is a challenging problem in signal processing. The noise attenuation performance of feedforward active noise control (ANC) relies on the preciseness of a reference signal of a primary noise field. To capture the precise reference signal for controlling a sizable space, a large number of reference microphones are required, which reduces system viability. In this study, we exploit an efficient representation of the reference signal in spherical harmonic (SH) domain by utilizing the inherent sparseness of the noise field. The main contributions of this work are as follows. (1) A general reference microphone geometry can be used. The implementation difficulty in the array structure, which is recognized as the common issue of SH-domain signal processing, e.g., use of a fully surrounding spherical array, is reduced by using the fields translation based on the addition theorem. (2) The accuracy of low-frequency signal decomposition is improved. The low accuracy of low-frequency signal decomposition in compressive sensing (CS), which is commonly reported in the literature, is improved by applying signal representation in SH domain. (3) System robustness is increased. The robustness of the system is increased by considering a noise source spatial distribution of both the interior and exterior sound fields, which is not possible in the case of a general signal representation in SH domain. Experimental results indicate that the noise attenuation performance of our proposed method exceeds that of existing solutions. The flexibility of the array structure is also increased, which leads to a more feasible practical system setup.

The aim of spatial active noise control (ANC) is to attenuate noise over a certain space. Although a large-scale system is required to achieve spatial ANC, mode-domain signal processing makes it possible to reduce the computational cost and improve the performance. A higher-order source (HOS) has an advantage in sound field control due to its controllable directivity patterns. An array of HOS can suppress an undesired exterior sound propagation while occupying a smaller physical space than a conventional omnidirectional loudspeaker array. In this paper, we propose two types of adaptive algorithm for spatial ANC using HOSs, which provide a trade-off between efficiency and error robustness against loudspeaker placements. Numerical simulations in a reverberant environment show the efficacy of the proposed algorithms compared with the conventional multipoint adaptive spatial ANC algorithm.

Active noise control (ANC) over a sizeable space requires a large number of reference and error microphones to satisfy the spatial Nyquist sampling criterion, which limits the feasibility of practical realization of such systems. This paper proposes a mode-domain feedforward ANC method to attenuate the noise field over a large space while reducing the number of microphones required. We adopt a sparse reference signal representation to precisely calculate the reference mode coefficients. The proposed system consists of circular reference and error microphone arrays, which capture the reference noise signal and residual error signal, respectively, and a circular loudspeaker array to drive the anti-noise signal. Experimental results indicate that above the spatial Nyquist frequency, our proposed method can perform well compared to a conventional methods. Moreover, the proposed method can even reduce the number of reference microphones while achieving better noise attenuation.

This paper proposes a technique for adding more prosodic variations to the synthetic speech in HMM-based expressive speech synthesis. We create novel phrase-level F0 context labels from the residual information of F0 features between original and synthetic speech for the training data. Specifically, we classify the difference of average log F0 values between the original and synthetic speech into three classes which have perceptual meanings, i.e., high, neutral, and low of relative pitch at the phrase level. We evaluate both ideal and practical cases using appealing and fairy tale speech recorded under a realistic condition. In the ideal case, we examine the potential of our technique to modify the F0 patterns under a condition where the original F0 contours of test sentences are known. In the practical case, we show how the users intuitively modify the pitch by changing the initial F0 context labels obtained from the input text.

This paper describes an approach to HMM-based expressive speech synthesis which does not require any supervised labeling process for emphasis context. We use appealing-style speech whose sentences were taken from real domains. To reduce the cost for labeling speech data with an emphasis context for the model training, we propose an unsupervised labeling technique of the emphasis context based on the difference between original and generated F0 patterns of training sentences. Although the criterion for the emphasis labeling is quite simple, subjective evaluation results reveal that the unsupervised labeling is comparable to the labeling conducted carefully by a human in terms of speech naturalness and emphasis reproducibility. Index Terms: HMM-based speech synthesis, expressive speech, emphasis expression, unsupervised labeling, F0 generation