Hareo Hamada

Fast deconvolution of multichannel systems using regularization

A very fast deconvolution method, which is based on the fast Fourier transform (FFT), can be used to control the outputs from a multichannel plant comprising any number of control sources and error sensors. The result is a matrix of causal finite impulse response filters whose performance is optimized at a large number of discrete frequencies. The paper is particularly aimed at multichannel sound reproduction and more specifically reproducing the sound field from a set of loudspeakers.

An adaptive noise control system in air‐conditioning ducts

This paper discusses an adaptive control system for the active cancellation of acoustic noise in air‐conditioning ducts. Recently, in several systems for active noise control, efforts have been focused on the use of adaptive digital filters to implement the system controller used to produce the artificial sound. The approach used in the newly proposed model assists in canceling broadband noise and also helps in improving the adaptation speed for colored noise. This model consists of a system identification (IDT) process and an adaptive noise cancellation (ANC) process. Two stand‐alone types of adaptive controllers using the fast‐least‐mean‐squares (FLMS) algorithm and the variable‐step least‐mean‐squares (VS‐LMS) algorithm were used in various experiments. Results of experiments in actual air‐conditioning ducts are presented.

Audio device and method for generating surround sound having first and second surround signal generation units

An audio device capable of easily generating two or more sets of surround signals based on 2-channel stereo signals and a method for generating surround sound are provided. The audio device 100 includes: an SL signal generation section 20 and a BL signal generation section 40 as first surround signal generation units for receiving an L signal and an R signal as 2-channel stereo signals, extracting a component of the R signal having high correlation with the L signal, subtracting the component from the L signal, thereby generating a first surround signal; and an SR signal generation section 30 and a BR signal generation section 50 as second surround signal generation units for extracting a component of the L signal having high correlation with the R signal, subtracting the component from the R signal, thereby generating a second surround signal. The level of subtracting a component from the L signal or the R signal for generating the first or second surround signal is differentiated each other between the plural sets.

Audio device and playback program for the same

The input signal X of one channel is divided by a multi-stage delay processing device Z−1 and each of the outputs is superimposed by a specified coefficient by a coefficient processing device W0, W1, . . . , Wk. The results are added by an adder, thereby providing a correlation eliminating filter for extracting a signal component from the input signal X of one channel having a high correlation with the input signal Y of the other channel. There is provided a coefficient updating processing device 5 for successively changing the feature of the correlation eliminating filter according to an error signal e obtained from the output signal RES and the input signal Y from the other channel, and the input signal X of one channel. A surround signal is obtained from a difference between the output RES from the correlation eliminating filter and the input signal Y of the other channel. Thus, upon reproduction of a two channels stereo signal, it is possible to generate a surround signal not giving uncomfortable feeling to a listener.

Subjective investigation of a new sound reproduction system (stereo dipole)

Recently, a new transaural sound reproduction system, referred to as a ‘‘stereo dipole’’ (SD) reproduction system, has been proposed, in which a closely spaced pair of loudspeakers is located in front of a listener. Theoretical investigation was carried out by computer simulations and it was reported that the SD system would be capable of providing a relatively large equalized area to the listener and robustness with respect to a listener’s head movement compared with those given by the standard loudspeaker arrangements. This paper deals with a subjective investigation of the SD system. It is known that the ordinary transaural system has a disadvantage in the quality of reproduced sound; that is, the reproduction of unnatural and antiphase virtual sound from the psychoacoustic point of view. Therefore, subjective listening tests were carried out, which aimed to compare the SD system with the ordinary system. The significance of the SD system on reproduced sound perception is suggested.

A study of pinna effect on head‐related transfer functions

For the sound localization, an interaural difference is important. To obtain the desired localization, it is necessary to use a head‐related transfer function (refer to HRTF), which is defined as the transfer function between a sound source and a surface of the ear drum. In our previous work, as far as in the case of reproducing the virtual sound source in the horizontal plane, it is confirmed to be capable of remarkable localization by using the numeric data calculated by a rigid sphere head model instead of a dummy head microphone. However, it is necessary to solve the problem of the front back confusion and the sound localization of the vertical plane. For that, it might be essential to add the pinna information to the rigid sphere model, to examine how each part of the pinna is related to the localization.

Evaluation of the subjective effects for active noise control

Active noise control (ANC) is a technique which can reduce low‐frequency noise effectively. The physical effects of an ANC system have been studied by many researchers; however, it is also requested to evaluate ANC systems from the viewpoint of the psychological effects on listeners. On the other hand, general noise assessments have been carried out by the equivalent noise level criteria such as the LAeq. Indeed, the LAeq is a well‐designed judgment criterion; however, the evaluation using the LAeq criterion includes an A‐weighted filtering. This may cause problems to occur when we treat noise that contains a lot of low‐frequency noise components rather than high‐frequency components. That is to say, even if the equivalent sound level is small, the low‐frequency noise may annoy, and cause harmful effects to, the listener in different ways than the high‐frequency noise. The purpose of this paper is to evaluate the effects of an ANC system on human hearing by carrying out a series of experiments. The subjec...

Design of a filter matrix used for stereo dipole transaural systems

A new transaural system, referred to as the ‘‘stereo dipole’’ reproduction system (hereafter the SD system), using a closely spaced pair of loudspeakers in front of a listener, has been introduced recently. In this paper, the deconvolution techniques used in the SD system, in which a 2×2 matrix of digital filters is used to compensate for the response of the loudspeakers and to create virtual sound sources, are dealt with. The deconvolution method is based on the analysis of a matrix of exact least‐squares inverse filters. Two kinds of design techniques are developed. One is a very fast FFT‐based deconvolution method; the other is a time‐domain counterpart to the FFT method. The well‐known principles of least‐squares optimization and regularization are introduced in both the design methods. The relationships between the characteristics of the deconvolution filters and loudspeaker arrangement are investigated. In particular, the theoretical analysis is concerned with the locations of poles and zeros and th...

Binaural sound reproduction in a stereo dipole system

A new transaural system referred to as a ‘‘stereo dipole’’ reproduction system using a closely spaced pair of loudspeakers in front of a listener has been introduced recently. Computer simulations using theoretical models, in which the listener’s head is assumed to be a perfectly rigid sphere, revealed that sound field equalization by the SD system including inverse filtering and also for the virtual source reproduction can be achieved in a wide range of area relative to the standard loudspeaker arrangements. This paper will deal with a further investigation of the SD system in terms of the controlled acoustic field around the head. The theoretical model used in the computer simulation is extended so that one can calculate an impulse response at any point while considering the influence of a rigid sphere. This model is used to compare the sound field around the spherical obstacle due to the SD system with one by the ordinary transaural system.

Quantitative evaluation of hypernasality in cleft palate patients

To quantitatively evaluate hypernasality in cleft palate patients, the Japanese vowel /i/ pronounced by six cleft palate patients and four normal children (controls) of similar ages was analyzed acoustically by cepstrum analysis. Spectrum envelopes obtained by the cepstrum method were evaluated every 1/3 octave to obtain the mean level in each band. Ten listeners evaluated a speech sample from each subject for degree of nasality on an equal interval scale ranging from 0 (no nasality) to 4 (strongest nasality). Two factors were obtained from the factor analysis of the judged scores. The first factor, which accounted for 77% of the total variance, was the consensus perception of nasality. The second factor, which accounted for 9%, was the difference among the individual listeners. The levels in two 1/3 octave bands were highly correlated with the first factor. The central frequencies of these two bands were 1 and 5 kHz.