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37 巻, 1 号
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吉川 昭吉郎原稿種別: 本文
1980 年 37 巻 1 号 p. 1-
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーPDF形式でダウンロード (206K) -
桑原 尚夫原稿種別: 本文
1980 年 37 巻 1 号 p. 2-10
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーA perceptual experiment is performed for two forms of non-stationary synthetic vowels. First, three-vowel sequences of the form /uVu/ are synthesized by a terminal-analog speech synthesizer as the test stimuli. The second formant frequency of the middle vowel is changed along a stimulus continuum raging from /u/ to /i/, while other formant frequencies are held constant. Two parameters, the rate of formant transition and the duration of the stationary part of the middle vowel, are taken into account as the possible factors that may have effects on vowel perception. A significant shift in the perceptual phoneme boundary between /u/ and /i/ for the middle vowels is found to occur owing to the surrounding vowels /u/. But, neither of the two parameters described above has a large effect on the vowel identification. Second, two-vowel sequences of the forms /uV/ and /Vu/ are used as the test stimuli of find out which vowel, preceding, has a more important influence on the vowel perception in terms of boundary shift. The results reveal that the perceptual phoneme boundary between /u/ and /i/ for the vowel /V/ in the form /Vu/ shifts to a considerable extent towards the /u/ area compared with that of /uV/. This indicates that the succeeding vowel has a more important influence that the preceding one on the phoneme identification of a vowel in connected speech.抄録全体を表示PDF形式でダウンロード (1151K) -
春日 正男原稿種別: 本文
1980 年 37 巻 1 号 p. 11-18
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーThe study of audio signal processing for high fidelity transmission is making a great progress as digital audio technology is prevailing. In particular, digital filters have found their effective applications where band-limiting is necessary. Whereas variable attenuation equalizers in digital design having an arbitrary frequency characteristics are often demanded. This paper describes a design of digital filters for variable attenuation equalizers. Initially, the conditions for the realization of the equalizer and the implementation of digital filter are studied in a form of simple hardware, which lead an adoption of infinite impulse response (IIR) digital filters. The transfer functions in the continuous system and in the discrete system are compared and investigated. The design methods in the continuous system for equalization is proved to be applicable to discrete systems. The three design parameters and the methods for varying frequency characteristics are proposed and examples of actual equalizers are shown. Transformation errors incurred with regard to the sampling frequencies are discussed. This approach to variable attenuation equalizers may be expected to open a new field of digital filters.抄録全体を表示PDF形式でダウンロード (920K) -
佐々木 良平原稿種別: 本文
1980 年 37 巻 1 号 p. 19-25
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーThe sound propagating in an acoustically lined duct of an aeroengine is incident to the acoustic treatment with oblique angle of incidence, rather than normal to the surface. Therefore, the frequency band where the largest absorption can be obtained changes from that calculated in the assumption that the sound would propagate perpendicularly against the surface. So the specification of acoustic treatment should be determined in terms of impedance in order to maximize noise attenuation at a certain frequency range. The impedance varies with frequency, kind of modes, main flow Mach number and duct geometries. In this paper, the relation between the impedance of the duct wall and the noise attenuation is analyzed, and acoustic test using a turbofan engine is conducted to evaluate the feasibility of the design technique of acoustic treatment.抄録全体を表示PDF形式でダウンロード (779K) -
河原 英紀, 筧 一彦原稿種別: 本文
1980 年 37 巻 1 号 p. 26-32
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーThis paper describes a new algorithm for estimating speech loudness function (dependency of each frequency component on total speech loudness) which is suitable when applied to objective rating assessments on loudness for telephoneconnections. This algorithm also provides a more efficient and reliable calculation method than the previous one. Loudness differences between reference system (NOSFER) and telephone systems are calculated from a 2-step model which consists of : i) non-linear transformation from sensitivity differences to psychometric differences and ii) weighted accumulation of psychometric differences. Model parameters are determined so as to minimize modified PSS (Prediction Sum of Squares), a criterion for selecting meaningful variables. Loudness balancing tests on various filters are carried out to examine this algorithm. Filters employed are implemented as FIR digital filters on a mini-computer software. Experimental results show that a few independent parameters can provide good agreement between computed and measured values and a parameter for nonlinearity has approximately the same value as the conventional power index of loudness.抄録全体を表示PDF形式でダウンロード (655K) -
時田 保夫原稿種別: 本文
1980 年 37 巻 1 号 p. 33-36
発行日: 1980/12/25
公開日: 2017/06/02
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樋口 重光, 三瓶 徹原稿種別: 本文
1980 年 37 巻 1 号 p. 37-40
発行日: 1980/12/25
公開日: 2017/06/02
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今泉 敏原稿種別: 本文
1980 年 37 巻 1 号 p. 41-44
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーPDF形式でダウンロード (517K) -
西村 実原稿種別: 本文
1980 年 37 巻 1 号 p. 45-
発行日: 1980/12/25
公開日: 2017/06/02
ジャーナル フリーPDF形式でダウンロード (191K)
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