• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.352-356
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• 2010

In this study, it is discussed how the directional property of the sound lateralization is influenced by proceeding or succeeding tone. The acoustic source applied here is a reference sound which has 0.5 msec interaural time difference(ITD). Based on this reference sound, interfering sounds with five levels of magnitude are applied to the subjects with four kinds of inter-stimuli time intervals(ISI). The interfering sounds are also added as two different types, proceeding tone and succeeding tone. Additionally, in order to investigate a frequency influence, the reference sound and the interfering sounds are generated by using 2kHz, 4 kHz and a white noise. As a result, the influence on lateralization by proceeding tone is lager than that by succeeding tone. It can consider this result as the effect of temporal masking on lateralization. Moreover, there are small differences of masking effect on lateralization by combinations of pure tone. This result shows that the dependency of frequency domain between reference sound and interfering sound is small on the sound lateralization.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.268-271
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• 2008

A sound-masking technique was applied to the urban public spaces with multiple noises in order to reduce the annoyance of the construction and traffic noise. In addition, the effects of masking system were investigated in order to improve the soundscape. In several urban spaces, building construction and nearby road traffic noise were recorded using a head and torso simulator and sound quality (SQ) system which specifies the spectral and temporal aspects of the noises. The sound-masking system which consists of the distributed speakers and control devices was applied to the boundaries of the construction sites. Synthesized masking sounds were produced with consideration of SQ characteristics of the multiple noise. Subjective evaluations on the soundscape were conducted to verify effectiveness of the system under the conditions with and without the masker sounds.

• Journal of the Korea Society of Computer and Information
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• v.29 no.6
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• pp.89-100
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• 2024

The primary goal of software testing is to identify and correct errors within software. A key challenge in this process is error masking, where errors disappear internally before reaching the output. This paper investigates the causes and characteristics of error masking, which complicates software testing. The study involved injecting artificial errors into three software programs to examine the extent of error masking by various test cases and to explore the underlying reasons. The experiment yielded four major findings. First, about 50% of the error masking occurred because the errors were not executed. Second, among various operators, logical and arithmetic operators masked errors less frequently, while relational and temporal operators tended to mask errors more extensively. Third, certain test cases demonstrated exceptional effectiveness in propagating errors to the output. Fourth, the type of error injected influenced the masking effect.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.2
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• pp.231-237
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• 2008

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.91-98
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• 2005

In this paper, we propose a temporal error concealment method based on the edge adaptive masking. In the method, four regions around the corrupted block - top, bottom, left, and right - are defined and the edge features of the regions are extracted by applying an edge operator for each direction. The size of a mask for the boundary matching is determined by the edge information, which can be considered as a criterion to measure the activity of the boundary region. In other words, it is determined such that the size of the mask is proportional to the amount of edge-component extracted from each region in order to yield the higher reliability on boundary matching. This process is equivalent to applying weights depending on the edge features, which leads the improved motion vector. In experiments, it is verified that the proposed method outperforms the conventional methods in terms of image quality, and then its merits and demerits are discussed.

• The Korean Journal of Applied Statistics
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• v.28 no.6
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• pp.1275-1288
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• 2015

A microscale weather analysis module (about 1km or less) is a microscale numerical weather prediction model designed for operational forecasting and atmospheric research needs such as radiant energy, thermal energy, and humidity. The accuracy of the module is directly related to the usefulness and quality of real-time microscale weather information service in the metropolitan area. This paper suggests an object based verification method useful for spatio-temporal evaluation of the accuracy of the microscale weather analysis module. The method is a graphical method comprised of three steps that constructs a lattice field of evaluation statistics, merges and identifies objects, and evaluates the accuracy of the module. We develop lattice fields using various evaluation spatio-temporal statistics as well as an efficient object identification algorithm that conducts convolution, masking, and merging operations to the lattice fields. A real data application demonstrates the utility of the verification method.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.70-76
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• 2009

In speech communications in noisy environments, speech intelligibility is seriously degraded due to the masking effect of ambient noise. In this paper, a new method to improve speech intelligibility in noisy environments is proposed. Based on the perception theory that the temporal envelope plays a major role in determining intelligibility, the proposed method uses a novel operation that enhances the fluctuation of band-wise temporal envelope and also contains pitch enhancement for improving speech naturalness. In addition, a new subjective evaluation scheme employing binaural listening is proposed in order to measure more reliable performance. The subjective performance measured with the proposed scheme shows that the proposed method improves both intelligibility and naturalness in various environments, whereas a function parameter can control the performance trade-off between intelligibility and naturalness.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.65-70
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• 2000

It is blown that the loudness is one of the most important metrics in assessing the sound quality and a calculation method for loudness has been standardized for steady sounds. In this study, a new loudness model is suggested for dealing with the transient sound for a unified analysis of various practical sounds. A signal processing technique is introduced for this purpose, which is required for the band subdivision and the prediction of band-level change of transient sounds. In addition, models for the post-masking and the temporal integration are adopted in the analysis of the loudness of transient sounds. In order to solve the problem of the conventional loudness model in the pure-tone signal processing, a critical band filter is employed in the analysis, which consists of 47 critical filters having a filter spacing of a half of the critical bandwidth. For testing the effectiveness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.822-832
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• 1994

DSC-HDTV image compression algorithm removes spatial, temporal, and amplitude redundancies of an image by using transform coding, motion-compensated predictive coding, and adaptive quantization, respectively. In this paper, leak processing method which is used to recover image quality quickly from scene change and transmission error and adaptive quantization using perceptual weighting factor obtained by HVS are proposed. Perceptual weighting factor is calculated by contrast sensitivity, spatio-temporal masking and frequency sensitivity. Adaptive quantization uses the perceptual weighting factor and global distortion level from buffer history state. Redundant bits according to adaptation of HVS are used for the next image coding. In the case of scene change, DFD using motion compensated predictive coding has high value, large bit rate and unstabilized buffer states since reconstructed image has large quantization noise. Thus, leak factor is set to 0 for scene change frame and leak factor to 15/16 for next frame, and global distortion level is calculated by using standard deviation. Experimental results show that image quality of the proposed method is recovered after several frames and then buffer status is stabilized.