• Journal of the Ergonomics Society of Korea
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• v.23 no.3
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• pp.89-99
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• 2004

A comprehensive psychophysical scaling system, multiple-modality magnitude estimation system (MMES) has been designed to measure subjective multidimensional human perception. Unlike paper-based magnitude estimation systems, the MMES has an additional auditory peripheral cue that varies with corresponding visual magnitude. As the simplest, purely psychological case, bimodal divided-attention conditions were simulated to establish the superiority of the MMES. Subjects were given brief presentations of pairs of simultaneous stimuli consisting of visual line-lengths and auditory white-noise levels. In the visual or auditory focused-attention conditions, only the line-lengths or the noise levels perceived should be reported respectively. On the other hand, in the divided-attention conditions, both the line-lengths and the noise levels should be reported. There were no significant differences among the different attention conditions. Human performance was better when the proportion of magnitude in stimulus pairs were identically presented. The additional auditory cues in the MMES improved the correlations between the magnitude of stimuli and MMES values in the divided-attention conditions.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.7
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• pp.308-316
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• 2002

In this paper, we propose a robust digital copyright-protection technique based on the concept of human auditory system. First, we propose a watermarking technique that accepts the various attacks such as, time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC WMA. Second, we implement audio PD(portable device) for copyright protection using proposed method. The proposed watermarking technique is developed using digital filtering technique. Being designed according to critical band of HAS(human auditory system), the digital filers embed watermark without nearly affecting audio quality. Before processing of digital filtering, wavelet transform decomposes the input audio signal into several signals that are composed of specific frequencies. Then, we embed watermark in the decomposed signal (0kHz~11kHz) by designed band-stop digital filer. Watermarking detection algorithm is implemented on audio PD(portable device). Proposed watermarking technology embeds 2bits information per 15 seconds. If PD detects watermark '11', which means illegal song. PD displays "Illegal Song" message on LCD, skips the song and plays the next song, The implemented detection algorithm in PD requires 19 MHz computational power, 7.9kBytes ROM and 10kBytes RAM. The suggested technique satisfies SDMI(secure digital music initiative) requirements of platform3 based on ARM9E core.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.81-87
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• 2001

The human auditory system is appropriate for the "constant Q"system. The STFT (Short Time Fourier Transform) is not suitable for the auditory perception model since it has constant bandwidth. In this paper, the CWT (continuous wavelet transform) is employed for the auditory filter model. In the CWT, the frequency resolution can be adjusted for auditory sensation models. The proposed CWT is applied to the modeling of the JNVF. In addition, other signal processing methods such as STFT, VER-FFT and VFR-STFT are discussed. Among these methods, the model of JNVF (Just Noticeable Variation in Frequency) by using the CWT fits in with the JNVF of auditory model although it requires quite a long time.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.397-400
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• 2002

This paper has been studied speech enhancement method in noisy environment. By mean of that we prefer human auditory mechanism which is perfect system and applied wavelet transform. Multi-resolution of wavelet transform make possible multiband spectrum analysis like human ears. This method was verified very effective way in noisy speech enhancement.

• International Journal of Precision Engineering and Manufacturing
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• v.7 no.2
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• pp.41-45
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• 2006

This paper deals with human spatial cognition using visual and auditory stimulation. More specially, this investigation is to observe the relationship between the head and the eye motor system for the localization of visual target direction in space and to try to describe what is the role of right-side versus left-side pinna. In the experiment of visual stimulation, nineteen red LEDs (Luminescent Diodes, Brightness: $210\;cd/^2$) arrayed in the horizontal plane of the surrounding panel are used. Here the LEDs are located 10 degrees apart from each other. Physiological parameters such as EOG (Electro-Oculography), head movement, and their synergic control are measured by BIOPAC system and 3SPACE FASTRAK. In the experiment of auditory stimulation, one side of the pinna function was distorted intentionally by inserting a short tube in the ear canal. The localization error caused by right and left side pinna distortion was investigated as well. Since a laser pointer showed much less error (0.5%) in localizing target position than FASTRAK (30%) that has been generally used, a laser pointer was used for the pointing task. It was found that harmonic components were not essential for auditory target localization. However, non-harmonic nearby frequency components was found to be more important in localizing the target direction of sound. We have found that the right pinna carries out one of the most important functions in localizing target direction and pure tone with only one frequency component is confusing to be localized. It was also found that the latency time is shorter in self moved tracking (SMT) than eye alone tracking (EAT) and eye hand tracking (EHT). These results can be used in further study on the characterization of human spatial cognition.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.7 no.1
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• pp.45-50
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• 2013

Recently, a method for detecting animal tinnitus objectively, gap prepulse inhibition of acoustic startling (GPIAS) were reported. However, the GPIAS method is difficult to be directly applied to human tinnitus research because measuring repeatedly startling responses evoked by more than 110 dB SPL acoustic stimuli for human is not easy. In this paper, the auditory late latency response (ALLR) measurement system which can measure conveniently evoked potentials involving the information about the brain cortical activity related with auditory psychologic phenomena such as a tinnitus has been implemented. By using the implemented system, 8 persons with normal hearing sense have been experimented to measure N1-P2 amplitudes of ALLRs evoked by gap prepulse based acoustic stimuli. Through the experimental results, the prepulse inhibitions of all the participants' N1-P2 responses have been observed and their characteristics were evaluated. And it is verified that our implemented system can be utilized as a device for researching and evaluating the objective tinnitus detection method in the future study.

• Journal of Navigation and Port Research
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• v.37 no.4
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• pp.401-407
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• 2013

In parallel with the development of ship equipment, bridge systems have been improved, but marine accidents due to human error have not been reduced. Recently, research in nautical bridge equipment has focused on suitable ergonomic designs in order to reduce these errors due to human factors. In a bridge of a ship, there are numerous auditory signals that deliver important information clearly to the sailors. However, only a few studies have been conducted related to the human recognition of these auditory signals. There are three types of auditory signals: voice alarms, abstract sounds, and auditory icons. This study was conducted in order to design more appropriate auditory icons using a sensibility evaluation method. The auditory icons were rated to have five warning situations (engine failure, fire, steering failure, low power, and collision) using the Semantic Differential Method. It is expected that the results of this study will be used as basic data for auditory displays inside bridges and for integrated bridge alarm systems.

• Speech Sciences
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• v.5 no.1
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• pp.167-179
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• 1999

The hearing mechanism is a complicated system. Sound is generated by a source that sends out air pressure or power. The pressure or power makes the sound waves. These waves reach the eardrum, or tympanic membrane, which vibrates at a rate and magnitude proportional to the nature of the sound waves. The tympanic membrane transforms this vibration into the mechanical energy in the middle ear, which in turn converts it to the hydraulic energy in the fluid of the inner ear. The hydraulic energy stimulates the sensory cells of the inner ear which send neuroelectrical impulses to the central auditory nervous system. The passive perception of auditory information starts just here. The listener gives attention to the speech sound, differentiates the sound from background noise, and integrates his experience with similar sounds. The listener then puts all of these aspects of audition into the context of the moment to identify the nature of sound. This has a major role in human communication. This paper provides an overview of the nature and characteristics of sound, the structure and function of the auditory system, and the way in which sound is processed by the auditory system.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.10
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• pp.181-187
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• 2013

When there are background noises or some people speaking at the same time, a human's auditory sense has the ability to listen the target speech signal with a specific purpose through Auditory Scene Analysis. The CASA system with human's auditory faculty system is able to segregate the speech. However, the performance of CASA system is reduced when the CASA system fails to determine the correct position of the speech. In order to correct the error in locating the speech on the CASA system, voice activity detection algorithm is proposed in this paper, which is a combined auditory scene analysis with PAR(Periodic to Aperiodic component Ratio). The experiments have been conducted to evaluate the performance of voice activity detection in environments of white noise and car noise with the change of SNR 15~0dB. In this paper, by comparing the existing algorithms (Pitch and Guoning Hu) with the proposed algorithm, the accuracy of the voice activity detection performance has been improved as the following: improvement of maximum 4% at SNR 15dB and maximum 34% at SNR 0dB for white noise and car noise, respectively.

• Journal of Institute of Control, Robotics and Systems
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• v.16 no.5
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• pp.415-419
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• 2010

A variety of methods for sound source localization have been developed and applied to several applications such as noise detection system, surveillance system, teleconference system, robot auditory system and so on. In the previous work, we proposed the sound source localization using the spatially mapped GCC functions based on TDOA for robot auditory system. Performance of the proposed one for the noise effect and estimation resolution was verified with the real environmental experiment under the single source assumption. However, since multi-talker case is general in human-robot interaction, multiple source localization approaches are necessary. In this paper, the proposed localization method under the single source assumption is modified to be suitable for multiple source localization. When there are two sources which are correlated, the spatially mapped GCC function for localization has three peaks at the real source locations and imaginary source location. However if two sources are uncorrelated, that has only two peaks at the real source positions. Using these characteristics, we modify the proposed localization method for the multiple source cases. Experiments with human speeches in the real environment are carried out to evaluate the performance of the proposed method for multiple source localization. In the experiments, mean value of estimation error is about $1.4^{\circ}$ and percentage of multiple source localization is about 62% on average.