• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.58-65
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• 2001

Recently, a 5-degrees-of-freedom (DOF) reverberation model was proposed as a method of representing subjective perception of reverberation as objective measures[1]. This model approximates sound energy decay curve by five objective measures, widely used in which have been concert hall acoustics. However, it is note worthy that there can be infinite number of impulse responses which correspond to a selected 5-DOF reverberation model. There may exist some filters making very unnatural and unrealistic sound. In this paper, the limitation of the 5-DOF reverberation model when it is used as a filter design criteria is investigated. When a 5-DOF reverberation model is given, additional constraints to get natural reverberation are suggested. This is based on the listening tests for several quite different source sounds.

• Archives of design research
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• v.16 no.4
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• pp.121-128
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• 2003

This is the research to find out which one is the best for using manual among the 3 kinds of methods such as 'Text manual' and 'Animation I' and 'Animation II' which is made by Virtual Realities. Three kinds of methods have been experimented respectively. The manual for 'Animation I' adopt the motion video with basis sound and the additional comment and sound is added on the 'Animation II'. Every 3 manual has been studied and estimated by T-test and additional subjective estimation respectively, and the conclusions are as follows. The 1st answer is that 'Animation manual' is more easier than 'text manual', and the 2nd answer is that 'Animation II' is easier than 'Animation I'. Through post-interview and test, It is known that the animation manuals, which has been showing the multimedia, is more attractive than text manual.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.410-418
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• 2012

Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.369-377
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• 2018

This study investigated the effect of noise and reverberation on subjective measure of speech transmission performance for elderly person with hearing loss in residential space through listening test. Floor impact, road traffic, airborne, and drainage noise were employed as the residential noise, and several impulse responses were obtained through room acoustical computer simulation for an apartment building. Sound sources for the listening test consisted of residential noises and speech sounds for boh the young (the original sound) and the aged (the sound filtered out by filters with frequency responses of hearing loss of 65 years elderly person). In the listening test, subjects evaluated speech intelligibility and listening difficulty for the presented word ($L_{Aeq}$ 55 dB) at three noise levels ($L_{Aeq}$ 30, 40, 50 dB) and three reverberation times (0.5, 1.0, 1.5 s). Results showed that the residential space with noise level lower than equal to 50 dB ($L_{i,Fmax,AW}$) for jumping noise and 40 dB ($L_{Aeq}$) for road traffic, airborne, and drainage noise had speech intelligibility of 90 % and over and listening difficulty of 30 % and below. Speech intelligibility and listening difficulty for the aged sound source was shown to be 0 % ~ 5 % lower and 2 % ~ 20 % higher than those for the young sound source, respectively.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1E
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• pp.30-38
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• 1995

In this paper, we evaluated the performance of vocoders for three-way conferencing using signal summation technique in digital mobile communication network. The signal summation technique yields natural mode of three-way conferencing, in shich the mixed voice signal from two speakers are transmitted to a third person, though there has been no useful speech coding technique for the mixed voice signal yet. We established Qualcomm code term prediction (RPE-LTP) vocoders to provide three-way conferencing using signal summation techinique. In addition, as the conventional speech quality measures are not applicable to the vocoders for mixed voice signals, we proposed two kinds of subjective quality measures. These are the sentence discrimination (SD) test and the modified degraded mean opinion score (MDMOS) test. The experimental results show that the output speech quality of the VSELP vocoder is superior to other two.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.

• The Journal of the Acoustical Society of Korea
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• v.43 no.4
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• pp.400-405
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• 2024

When cavitation noise occurs in ship propellers, the level of underwater radiated noise abruptly increases, which can be a critical threat factor as it increases the probability of detection, particularly in the case of naval vessels. Therefore, accurately and promptly assessing cavitation signals is crucial for improving the survivability of submarines. Traditionally, techniques for determining cavitation occurrence have mainly relied on assessing acoustic/vibration levels measured by sensors above a certain threshold, or using the Detection of Envelop Modulation On Noise (DEMON) method. However, technologies related to this rely on a physical understanding of cavitation phenomena and subjective criteria based on user experience, involving multiple procedures, thus necessitating the development of techniques for early automatic recognition of cavitation signals. In this paper, we propose an algorithm that automatically detects cavitation occurrence based on simple statistical features reflecting cavitation characteristics extracted from acoustic signals measured by sensors attached to the hull. The performance of the proposed technique is evaluated depending on the number of sensors and model test conditions. It was confirmed that by sufficiently training the characteristics of cavitation reflected in signals measured by a single sensor, the occurrence of cavitation signals can be determined.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.379-385
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• 2008

In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.5
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• pp.345-351
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• 2015

Sounds of electrically powered vehicle components such as window lift system, roller blind, etc., are required to be more comfortable and not to irritate the people emotionally. In this paper, a study was conducted to identify the significant sound quality metric and compose the luxuriousness sound quality index of electrically powered vehicle roller blind which is part of vehicle sunroof system. Before conducting subjective evaluation, sound characteristics of roller blind was analyzed and set the target operating sound for subjective evaluation. Thus, transfer sound of roller blind which has the characteristics of sound modulation was used for subjective evaluation. Multiple linear regression analysis was carried out by chosen Zwicker's metrics which are pointed by comments of jurors. Loudness and sharpness related metrics are prime metrics in luxuriousness sound quality index we composed. Also, effect of roller blind assay when it is attached to real vehicle was identified to evaluate the validity of index.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.38-45
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• 1995

In this paper a way to improve the performance of the long term prediction is proposed, which adopts the Multi-band Excitation (MBE) method in addition to the Code-Excited Linear Prediction (CELP) method at low bit rates below 4.8 kbps. In the proposed method, the multiband long term prediction is performed on the periodic components which still remain after the long term prediction of the conventional CELP method. At this point, the whole frequency region is divided into subbands whose size is equal to the spacing between the harmonics of the fundamental frequency, and the periodic multiband excitation signals. are represented as the sum of sine waves approximately as large as the spectrum of the excitation signals, so that the actual characteristics of the excitation signals can be better taken into account. To evaluate the performance of the proposed method, computer simulation is performed at 4.8 kbps. The 4.8 kbps DoD CELP and the 4.4 kbps IMBE were chosen as the reference vocoders for the speech quality measure. The result of the perceptual speech quality measure showed that the performance of the proposed method is better than that of the 4.8 kbps DoD CELP vocoder, and similar to that of the 4.4 kbps IMBE vocoder.