Toshio Sone

An optimum computer‐generated pulse signal suitable for the measurement of very long impulse responses

Transfer functions of acoustic systems often exhibit wide dynamic ranges and very long impulse responses. A ‘‘time‐stretched’’ pulse as proposed by Aoshima (ATSP), though originally given in a very specific form seems to be one of the most promising signals to measure transfer functions with characteristics of acoustic system mentioned as above. In this paper, this pulse (ATSP) is first generalized and then optimized for the measurement of long impulse responses. This optimized ATSP (OATSP) has an almost ideal characteristic to measure impulse responses shorter than its specific length N. Moreover, it is newly shown in this paper that OATSP has also a good characteristic to measure impulse responses longer than N. Discussion is presented on how to design OATSP suitable for a specific situation of measurement by analyzing errors, when the pulse is used to measure impulse responses longer than N.

Role of spectral cues in median plane localization

The role of spectral cues in the sound source to ear transfer function in median plane sound localization is investigated in this paper. At first, transfer functions were measured and analyzed. Then, these transfer functions were used in experiments where sounds from a source on the median plane were simulated and presented to subjects through headphones. In these simulation experiments, the transfer functions were smoothed by ARMA models with different degrees of simplification to investigate the role of microscopic and macroscopic patterns in the transfer functions for median plane localization. The results of the study are summarized as follows: (1) For front-rear judgment, information derived from microscopic peaks and dips in the low-frequency region (below 2 kHz) and the macroscopic patterns in the high-frequency region seems to be utilized; (2) for judgment of elevation angle, major cues exist in the high-frequency region above 5 kHz. The information in macroscopic patterns is utilized instead of that in small peaks and dips.

Adaptive feedback cancellation with frequency compression for hearing aids

The use of an adaptive feedback canceler (AFC) for howling suppression in hearing aids seems very attractive since it is not only unaffected by the changes in the operating environment, but it also limits signal degradation due to the feedback signal. This, however, requires a reference signal which is correlated with the feedback signal but not with the input signal. In hearing aids, such a signal is hard to obtain. The output signal could be used as reference if its correlation with the input signal could sufficiently be removed. If the reference signal is correlated with the input signal, the input signal will also be canceled by the AFC. Here, the use of a frequency compressor as a decorrelator is proposed. The performance of this system is then investigated via digital simulation. Results indicated that with the use of the proposed system and the proper choice of system parameters, an increase of about 18 dB in the howling margin could be achieved with minimal deterioration in output signal quality.

A new FFT algorithm of radix 3,6, and 12

A new algorithm for implementation of radix 3, 6, and 12 FFT is introduced. An FFT using this algorithm is computed in an ordinary (1,j) complex plane and the number of additions can be significantly reduced; the number of multiplication is also reduced. High efficiency of the algorithm is derived from the fact that, if an input sequence is favorably reordered, rotating factors can be treated in pairs so that the rotating factors are conjugate to each other.

Improvement of Hearing Ability by Directional Information

An investigation is undertaken of the ability to hear a signal sound in the presence of noise, when the signal and the interfering sound each have a directional characteristic and are separated from each other more than a certain degree, compared with the case when both are not separated. We attempt to determine through what process in our auditory system such a phenomenon occurs. We think that, when there are many sounds coming from many directions, directional information is extracted from the neural signals from both ears in our auditory system, each sound is localized in each place or direction, and a certain particular sound is selected and listened to. There is considerable difference, in the subjects ability to hear the sound, between the case in which a subject concentrates his attention on it, and the case in which he does not. For this interpretation, “attention” is introduced. It is clarified that the function that maintains the attention leads to the selective hearing of a signal relative to ...

Control of auditory distance perception based on the auditory parallax model

Abstract Simulation of the distance of a sound image within 2 m from a listener, in the absence of reflections and a loudness cue, was investigated. To do this, a model named the “auditory parallax model”, which focuses on the role of parallax angle information involved in head-related transfer functions (HRTFs), was examined with psychoacoustical experiments. For purposes of comparison, experiments were also done on an actual sound source, sound with digitally synthesized HRTFs, and sound with the interaural time difference (ITD) and interaural level difference (ILD) synthesized by the Hirsch–Tahara model. The perceived distance of a sound image by the actual sound source monotonically increased with the physical distance of the source up to 1–1.5 m without any cues of sound pressure level and reflections from walls. The perceived distance of a sound image simulated with the auditory parallax model and that with synthesized HRTFs showed tendencies very similar to those with the actual sound source. On the other hand, for a sound image produced by the Hirsch–Tahara model, the perceived distance of the sound image increased up to around 40 cm and then became saturated. These results show that simple synthesis of the ITD and ILD as a function of distance is insufficient to explain auditory distance perception. However, the results of the experiment in which HRTFs were simulated based on the auditory parallax model showed that the cues provided by the new model were almost sufficient to control the perception of auditory distance from an actual sound source located within about 2 m. Possible reasons for the good performance of the auditory parallax model are the resemblance between the relative frequency characteristics (shape) as well as the shape of ILD as a function of frequency simulated by the auditory parallax model and those of the actual HRTFs.

Auditory detection of frequency transition

The thresholds for the detection of frequency transition were obtained for the tone bursts in which the frequency was linearly changed from the initial value of 1000 Hz either during almost the whole burst duration or during a portion of it. The onset of transient had various delays from the beginning of the tone burst. The burst durations were 32, 102, 302, and 1000 msec. It was found that the nearly constant thresholds were obtained for a given burst duration as long as W(=T1+13T2) was held constant, where T1 was the duration of the initial segment of steady frequency and T2 was the duration of frequency transition. The thresholds were greater when the transition occurred near the burst onset than when it occurred near the cutoff, and in the former case, the thresholds were influenced by a rise time of tone burst. The thresholds were smallest when the frequency transitions occurred in the middle of the tone burst. It is suggested that there are two mechanisms for auditory detection of frequency transiti...

A digital hearing aid that compensates loudness for sensorineural impaired listeners

A method of designing a digital hearing aid suitable for sensorineural hearing loss with a narrow audible range is proposed. The input signal is divided into short-time blocks (8 ms), and its spectrum is calculated using short-time fast Fourier transforms. The frequency-gain characteristic of the block is determined using the calculated spectrum and loudness compensation functions which describe the relation between the loudness perceived by normal listeners and that perceived by an impaired listener. The input signal within the block is processed by a single digital filter using the calculated gain characteristics. This algorithm is realized using a digital signal processor. The results of monosyllabic speech tests were that the scores of the speech test for the proposed system were improved over a wide range of input levels compared with those for the linear system for nine of 13 sensorineural impaired subjects.<<ETX>>

On the Perception of Direction of Echo

The minimum time differences over which same two sounds coming from different directions in succession are heard separately in direction are obtained. When two clicks are used as stimuli, the second click is lateralized separately from the first click if the former lags more than 10 msec behind the latter and the second click is less than 20 dB smaller as compared with the first one. If an accessory click is inserted between these two clicks, the minimum time difference over which two stimuli are separately lateralized is longer than that in the case without an accessory click and its value amounts to 20∼30 msec. Such a minimum difference is 100∼200 msec in the case that the space between the first click and the second one is filled up with some stimuli. As one of the results, it is shown that this time difference is smaller for a sound containing more high‐frequency component than for one containing less. Moreover, when two sounds with same duration are given to each ear with an interaural time differenc...

Monaural phase effects on timbre of two‐tone signals

The effects of the phase difference between components of a two‐tone signal whose components have a frequency ratio of f2/f1=2.0 on its timbre were investigated. In the experiments, the similarities among six signals were judged with the complete method of triads. The subjective space in which the configuration of points with the different phase conditions lay was derived from the data by means of Torgerson’s multidimensional scaling program. An early study stated that the subjective space was a circle. Present results, however, revealed that the subjective space becomes one‐dimensional when either the amplitude ratio between components was large or when the frequency of the signal was high. This one‐dimensional configuration is interpreted in terms of the partially masked loudness of the higher frequency component of the signal.