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This study tested the validity of applying a signal detection model to Japanese listeners’ /r/–/l/ labeling behavior in a one‐interval identification task, and the hypothesis that Japanese listeners estimate each token’s goodness of fit to Japanese /r/ in labeling it as /r/ or /l/. The identification data obtained by manipulating subjects’ response criteria fit the signal detection model. The performance in the identification task was highly predictable based on the goodness of fit judgment, indicating that category goodness information contained in /r/ and /l/ is utilized in deriving underlying sensory decision variables. The data, however, also suggested that these variables are not a direct consequence of the goodness of fit judgment. Various implications of the signal detection model in the investigation of Japanese perception of /r/ and /l/ are discussed.

Most techniques for suppressing grating lobes in phased arrays while relaxing the interelement spacing requirement involve redistributing array elements in sparse aperiodic patterns, or varying the transmit-receive beam patterns. An alternative is presented which uses oversized array elements to eliminate grating lobes as a direct consequence of the element shape. It is shown that by using carefully shaped, overlapping elements, maximum scan angle can be exchanged for a reduced interelement spacing requirement.

Improving the validity of speech-recognition models requires an understanding of the conditions in which speech is experienced in everyday life. Listening conditions leading to a degradation of the signal—noise, competing talkers, disordered speech—have received most of the attention in that literature. But what about adverse conditions that do not alter the integrity of the signal, such as listening to speech under a non-auditory cognitive load (CL)? Drawing upon a variety of behavioral methods, this presentation investigates the effects of a concurrent attentional or mnemonic task on the relative reliance on acoustic cues and lexical knowledge during speech-perception tasks. The results show that listeners under CL downplay the contribution of acoustic detail and increase their reliance on lexical-semantic knowledge. However, greater reliance on lexical-semantic knowledge under CL is a cascaded effect of impoverished phonetic processing, not a direct consequence of CL. Ways of integrating CL into the fu...

A statistical‐decision theory based model for predicting speech intelligibility is introduced. This model, which we call the speech‐recognition sensitivity (SRS) model, aims at predicting speech‐recognition performance from the long‐term average speech spectrum, the masking excitation in the listener’s ear, the linguistic entropy of the speech material, and the number of response alternatives available to the listener. Unlike articulation‐index (AI) models, the SRS model can account for synergetic and redundant interactions among spectral bands of speech. It also accounts for effects of linguistic entropy and number of response alternatives on intelligibility scores without resorting to the empirically determined ad hoctransformations employed by AI models. The effects of linguistic entropy are modeled by an entropy‐dependent central noise, which modifies the listener’s identification sensitivity to the speech. The effect of the number of response alternatives on the test score is a direct consequence of using statistical decision theory. The SRS model also appears to predict how the effect of linguistic entropy varies with the filter condition and how linguistic entropy and language proficiency interact with the signal‐to‐noise ratio. Fits of the SRS model to data from the literature and consonant‐discrimination data collected in our laboratory will be presented. [Work supported by NIH and SigmaXi.] a)Current affiliation: GN ReSound Corporation.

The nonlinearity of the basilar membrane motion observed by Rhode can be accounted for by assuming that the linear losses (proportional to the membrane velocity) are supplemented by nonlinear losses—particularly losses proportional to the velocity raised to the third power. In this paper, the consequences of such a nonlinear‐loss law for the combination tone 2f1−f2 (the lower ’’cubic difference tone,’’ DCT) are examined. In agreement with experimental data, the amplitude of the CDT is found to approach a cube‐root dependence on the primary stimulus amplitudes when the two primary amplitudes are varied proportionally. When one primary amplitude is held fixed, the calculated CDT‐amplitude goes through a maximum when the other primary amplitude is increased—again as observed experimentally. In the model, this noteworthy amplitude behavior is a direct consequence of the cube‐root dependence of the membrane velocity on pressure for large pressures which has an amplitude limiting action on the generation of the...

The effect of changes in source position on the attenuation by a lined duct segment has been analyzed. The results show that variations in source position produce significant changes in the attenuation. This is the direct consequence of modifications in the model content of the exciting pressure field.Subject Classification: 20.45, 20.35.

In the history of speech research, sound spectrographs are playing the role of an almost invaluable tool. Besides the many advantages of these instruments, some of their limitations have also become evident. Quite a few of these are a direct consequence of the utilization of analog techniques in these devices. The present paper tries to show that digital‐processing techniques allow to implement a spectrographic system much more powerful than its analog counterpart. Its hardware consists of a general‐purpose computer adapted to accept digitally encoded speech signals, and a standard chain printer. The short‐time power spectra are computed utilizing the fast Fourier transform algorithm. Spectrograms with eight gray levels can be generated by multiprinting adequate sequences of characters out of the basic 48‐character set. The necessary computer programs are sufficiently simple to be easily tailored to the specific needs of any particular investigation. Wherever digital processing is used in speech research, the approach described exhibits its greatest advantages. The digital spectrograph can then be seen to constitute an organic part of the total software system being a subroutine that may be called whenever needed.